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International Journal of Multidisciplinary Approach and Studies

Phonocardiogram Signal Processing Using LMS Adaptive Algorithm

Article · April 2016

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 International Journal of Multidisciplinary Approach
and Studies ISSN NO:: 2348 – 537X

Phonocardiogram Signal Processing Using LMS Adaptive

Algorithm

M.Danida Deborah*, J.Prasad**,A.Aamina***,& A.Renuga Devi****,

* Student, Biomedical Engineering, Sri Ramakrishna Engineering college, Coimbatore,

** Assistant Professor (Sr.G), Biomedical Engineering, Sri Ramakrishna Engineering college, Coimbatore,
*** Student, Biomedical Engineering, Sri Ramakrishna Engineering college, Coimbatore,
**** Student, Biomedical Engineering, Sri Ramakrishna Engineering college, Coimbatore,

ABSTRACT

The heart sound is processed and displayed in the form of waveform called as
phonocardiogram (PCG) signal. The PCG signal can be obtained from electronic
stethoscope or from phonocardiograph is used as primary tool for diagnosis of heart valvular
diseases. There are two heart sounds: S1 known as "Lub" occurs due to the opening and
closure of Atreioventricular valve and S2 is known as" dub" sound occur due to opening and
closure of semilunar valve. The PCG signal acquired from electronic stethoscope or from
phonocardiograph contains environmental noise along with the heart sound. It is a challenge
to separate heart sound from noise. LMS are Adaptive finite impulse response filter, whose
coefficients or weights change over time to adapt to the statistics of a signal. LMS has low
computational complexity and it minimize the mean square value of the error signal. In this
work the random noise is correlated with PCG signal. The noisy PCG signal is processed for
noise cancellation using Least Mean Square (LMS) adaptive algorithm in MATLAB. The
PCG signal processed by LMS algorithm is measured based on the parameters namely
Minimum Mean Square Error (MMSE), computational complexity and peak signal to noise
ratio (PSNR).

Key Words: Phonocardiograph (PCG), Least Mean Square (LMS) algorithm and Matlab
software.

INTRODUCTION

The heart is an organ that pumps blood throughout the body via the circulatory system,
supplying oxygen and nutrients to the tissues and removing carbon dioxide and other wastes.
The tissues of the body need a constant supply of nutrition in order to be active. The heart has
four chambers: two upper chambers (the atria) and two lower ones (the ventricles). The right
side of the heart has impure blood and left side of the heart as pure blood. The tricuspid valve
and the mitral valve make up the atrioventricular (AV) valves, which connect the atria and
the ventricles [1]. The pulmonary semi-lunar valve separates the right ventricle from the
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pulmonary artery, and the aortic valve separates the left ventricle from the aorta. The
heartstrings, or chordae tendinae, anchor the valves to heart muscles. There are two
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circulatory system pulmonary and systemic circulatory system. In the pulmonary system,
deoxygenated blood leaves the right ventricle of the heart via the pulmonary artery and enter
the lungs for purification, oxygenated blood enter the left atrium of the heart via the
pulmonary vein. In the systemic circuit, oxygenated blood leaves the body via the left

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 International Journal of Multidisciplinary Approach
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ventricle to the aorta, and supplies body's tissues with oxygen. Deoxygenated blood returns
via veins to the venae cavae, re-entering the heart's right atrium.
Valves prevent backflow, keeping the blood flowing in one direction through the heart. Heart
valves can have three basic kinds of problems: regurgitation, stenosis, and atresia.
Regurgitation, or backflow, occurs if a valve doesn't close tightly. Blood leaks back into the
chambers rather than flowing forward through the heart or into an artery.
Stenosis occurs if the flaps of a valve thicken, stiffen, or fuse together [6]. This prevents the
heart valve from fully opening. As a result, not enough blood flows through the valve. Some
valves can have both stenosis and backflow problems.
Atresia occurs if a heart valve lacks an opening for blood to pass through [6].
Heart diseases is one of the major health problem thought the world. The recording of the
acoustic waves produced by the mechanical action of heart is known as phonocardiogram
signal (PCG). It is an effective tool for diagnosing heart valvular diseases. In real time,
acquired PCG signal contains noise along with it. It can be processed with LMS and NLMS
adaptive algorithm for effective noise cancellation.

HEART SOUND
Acoustic heart sounds are produced when the heart muscles open the valves to let blood flow
from chamber to chamber. Cardiac sounds are non-stationary sounds and in the low
frequency range, approximately 30-750Hz [4]. A normal heart will produce two heart sounds
(shown in Fig 1) S1 (Lub) and S2 (Dub). After that S3 is caused by the vibration of
ventricular valves. S4 occurs during the second phase of ventricular filling. The sounds are
produced due to blood flow in and out of the heart and movement of valves ie. AV valves and
semilunar valves [1].

Fig 1. Normal PCG signal

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First heart sound

S1 symbolizes the start of systole. The sound is created when the mitral and tricuspid valves
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close which allows the flow of blood from atria to ventricle after blood has returned from the
body and lungs. Frequency range is 30 Hz - 45 Hz [4]. S1 is high intensity, longer duration
and low frequency.

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 International Journal of Multidisciplinary Approach
and Studies ISSN NO:: 2348 – 537X

Second heart sound

S2 symbolizes the end of systole and the beginning of diastole. The sound is created when the
aortic and pulmonary valves close as blood exits the heart to the body and lungs which lie
with maximum frequency in the 50 Hz - 70 Hz [4] range with higher pitch.

Fig 2.

Abnormal PCG signal

Third (s3) and fourth (s4) heart sound

S3 (Fig 2) heard in children and patient with left ventricular failure due to rapid inflow of
blood from the atria into the ventricles. The duration is 0.1 to 0.2sec and the frequency is
below 30Hz [4]. It is produced by the contraction of the atria. It is not audible because it has
low amplitude and frequency. It occurs immediately before the first heart sound. The duration
is 0.03 to 0.06 second. Frequency range is 10-50Hz [4]. The S3 and S4 sounds are called
diastolic sounds and are generally inaudible in the normal adult but are commonly heard in
children.

DIGITAL FILTER
In signal processing, a digital filter is a system that performs mathematical operations on
a sampled, discrete-time signal to reduce or enhance certain aspects of that signal. Digital
filters are mainly used for two purpose:
Separation of signal that have been combined

Restoration of signal that have been distorted

Infinite Impulse Response (IIR)
IIR filters are one of two primary types of digital filter. The impulse response is "infinite"
because there is feedback in the filter. The system will be mostly unstable and computational
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complexity is high this is the main drawback of IIR filter.

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Finite Impulse Response (FIR)

FIR filters are one of two primary types of digital filters. If the single sample “1” is followed
by many “0”. Zeros will come after the sample”1” has made its way through the delay of the
filter. The advantages of FIR filters are FIR filters have an amazing property called Linear

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 International Journal of Multidisciplinary Approach
and Studies ISSN NO:: 2348 – 537X

phase [2]. It ensures that signals of all frequencies are delayed by the same amount of time,
thereby eliminating the possibility of phase distortion. It is mostly useful in audio
applications. FIR filters are always stable i.e. for a finite input, the output is always finite.

ADAPTIVE NOISE CANCELLATION

A filter which self adjust its transfer function is called adaptive filter (Fig 3). It required two
signals: input signal (signal and noise) and desired signal. The error signal is used as
feedback in the form of signal in adaptive filter which is used to refine the transfer function to
match changing parameters. Mostly the adaptive filters are used for reducing the noise
content.

Fig 3. Adaptive filter

Where (Fig 3)
x(n) – input signal
y(n) – output signal
d(n) – desired signal
e(n) – error signal

Least mean square algorithm

LMS is simplest algorithm used in adaptive filter because it calculates the filter coefficient
from the error signal [7]. It is a search algorithm; gradient vector computation is simplified by
modifying the objective function. It produces the least mean squares of the error signal which
is the difference between the desired and actual signal [5]. The standard LMS Algorithm
performs the following operations to update the coefficients of the adaptive filter (Fig 4)
Calculate the output signal y(n) from the adaptive filter. y(n) = w'(n)*x(n).
Calculate the error signal e(n) by e(n)=d(n)-y(n).
Update the filter coefficients by using the equation: w(n+1)=w(n)+µ*e(n)*u(n)
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Where µ = step size of the adaptive filter

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x(n) = input vector

w(n) = weight vector. (Assume the initial weight)
w(n) = filter coefficients vector
u(n)=filter input vector

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 International Journal of Multidisciplinary Approach
and Studies ISSN NO:: 2348 – 537X

Fig 4. Flow chart of LMS

SIMULATION USING MATLAB

The PCG audio signal is collected from the database “Michigan_Heart_Sounds”. The step
size chosen for simulation is mu = 0.002. The signal is sampled and then filtered using noise
cancellation technique.

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Fig 5. Noisy PCG signal

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 International Journal of Multidisciplinary Approach
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The audio signal was imported to Matlab by using wavread function. The first subplot of Fig
5 show the random noise generated and the second subplot (Fig 5) shows the addition of
random noise with the PCG signal.

Fig 6. LMS Output

The above Fig 6 is a result of LMS adaptive filter algorithm with the step size of 0.002. The
plot shows the LMS output signal and the error signal. The error signal is gradually
decreased; it signifies the efficiency of the LMS algorithm.

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Fig 7. LMS-Mean Square Error

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The above Fig 7 shows the MSE of LMS algorithm for the filter size from 10 to 50. The MSE
valve increase with increase in filter size. The small value of MSE indicates the performance

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of algorithm. The high values of MSE indicated that the initial start of the adaptive filter is
inadequate starting point to cause the adaptive filter to converge.

Fig 8. LMS – PSNR

The Fig 8 shows the Peak Signal to noise ratio based on LMS algorithm. The signal to noise
ratio should be high for better performance. In LMS algorithm the PSNR decrease with
increases in filter size. For LMS algorithm its better to have the small filter length for better
performance.

Filter Step
Algorithm MSE PSNR
size size

0.002 1.90e-002 17.21

16
0.025 4.95e-002 13.05
LMS
0.002 1.99e-002 17
25
0.025 4.60e-002 13.37

Table 1. Parameters Measured

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Table 1 infers the MSE and PSNR values for different filter size and step size. The increase
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in step size increases the MSE value and decreases the PSNR values. The same for increase
in filter size.

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 International Journal of Multidisciplinary Approach
and Studies ISSN NO:: 2348 – 537X

CONCLUSION

The Least Mean Square algorithm is simple to compute and implement. Computational
complexity and adaptation rate of LMS is low. The performance of the algorithm is measured
from the error signal, mean square error value and PSNR valve. The LMS algorithm,
minimize the mean square value of the error signal. The step size determines the convergence
speed of the LMS algorithm. Smaller the step size and smaller filter length better the
performance of LMS algorithm in noise cancellation of PCG signal.

REFERENCES

i. Elizabeth M. Cherry and Flavio H. Fenton “Heart Structure, Function and

Arrhythmias”, http://thevirtualheart.org/3dpdf/Heart_3d.pdf
ii. https://www.minidsp.com/applications/dsp-basics/fir-vs-iir-filtering
iii. Divya, Preeti Singh, Rajesh Mehra “Performance Analysis of LMS & NLMS
Algorithms for Noise Cancellation” International Journal of Scientific Research
Engineering & Technology (IJSRET) Volume 2 Issue 6 September 2013 ISSN 2278 –
0882
iv. http://enhanceedu.iiit.ac.in/wiki/images/PCG_BD.pdf
v. Acoustic echo cancellation using digital signal processing.pdf
vi. http://www.clevelandclinic.org/SEM_info/heartcenter/Valve_Disease/img/CCF_Valv
e_Disease_Guide.pdf
vii. Jyoti dhiman, shadab ahmad and kuldeep gulia” Comparison between Adaptive filter
Algorithms (LMS, NLMS and RLS)” International Journal of Science, Engineering
and Technology Research (IJSETR) Volume 2, Issue 5, May 2013. ISSN: 2278 –
7798

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